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msc at freeswitch.org Guest
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Posted: Tue Dec 09, 2008 9:53 am Post subject: [Freeswitch-users] Proto specific hangup cause issue |
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|
Thanks for your feedback. It definitely helps to know not only what
you need FS to do but why you need it to do so.
Do you have FS in production right now? Just curious.
Thanks,
MC
On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
<regs@kinetix.gr> wrote:
Quote: | "I already added 2 patches for you right. Just be clear about what you
want."
And I am grateful of that.
"it is protocol neutral, that's why it starts with sip_"
I didn't know that. I thought that the sip_ variables are protocol specific.
So one would expect there to be an iax_hangup_disposition,
woomera_hangup_disposition etc?
"Maybe you should beat around the bush less with your "requirements" for
your application you are expecting me to support for you."
I am just trying to gather statistics for my providers as I would with any
VoIP softswitch. (hangup causes per terminator per destination)
I don't think that this is a specific "application" rather than a general
necessity for VoIP carriers. It is also very useful for troubleshooting
purposes : when I look at my CDRs to find a call that I got a complain for,
I want to be able to tell if it was me or the provider who
hanged up and gave a specific hangup cause, so that I can troubleshoot the
issue better.
"Just be clear about what you want."
I want FS to reach that level of detailing and maturity in all aspects so
that it could be the softswitch of choice by any VoIP entrepreneur
(or hobbyist) and it is my strong belief that this can only be done by the
community giving feedback to the programmers about what
they find useful or not (i.e. experience from real-life situations). The
patches that you made the last few days were not intended for
me exclusively but for anyone that will face the same situations using FS.
If you want the community to stop sending feedback about
features/improvements you may as well close down this mailing list or just
use it as an announcement board.
I wish I was a c programmer and get involved with the project actively. But
I am not. And as far as I can tell most of the registered users
in this list aren't either. So they only way we can help is by testing and
suggesting.
Anthony Minessale wrote:
it is protocol neutral, that's why it starts with sip_
the variable can be any of:
send_bye
recv_bye
send_cancel
send_refuse
using that value you can determine the information you asked. I answered
your specific question which was:
determining "which side hanged up". Maybe you should beat around the bush
less with your "requirements" for your application you are expecting me to
support for you.
I already added 2 patches for you right. Just be clear about what you want.
On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs@kinetix.gr>
wrote:
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org
pstn:213-799-1400
________________________________
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr
-------------------------------------------
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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regs at kinetix.gr Guest
|
Posted: Tue Dec 09, 2008 10:28 am Post subject: [Freeswitch-users] Proto specific hangup cause issue |
|
|
We are currently in the migration process from our
current system to a FS based setup. We are in the process of
adapting our billing and routing to FS. All the CDRs (and variables)
related issues that we have been discussing on this mailing list
come from the need to extract the same level of information from FS as
we do with our current closed source proprietary system. So, we
chose FS because of the versatility it provides in every aspect (event
handling, config implementation etc.) and we strongly believe that all
these additions/fixes would be beneficial to many potential FS users.
We are at your disposal for more details in case you need
more information about what exactly we are trying to do. Basically,
our approach is from the VoIP carrier's point of view rather than the
PBX user's/implementor's. So, the details that we asked to be introduced
to FS come from real life issues that we have faced during the last few years
with various platforms and troubleshooting experiences with other VoIP carriers.
Michael Collins wrote: Quote: | Quote: | Thanks for your feedback. It definitely helps to know not only what
you need FS to do but why you need it to do so.
Do you have FS in production right now? Just curious.
Thanks,
MC
On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
<regs@kinetix.gr> (regs@kinetix.gr) wrote:
Quote: | "I already added 2 patches for you right. Just be clear about what you
want."
And I am grateful of that.
"it is protocol neutral, that's why it starts with sip_"
I didn't know that. I thought that the sip_ variables are protocol specific.
So one would expect there to be an iax_hangup_disposition,
woomera_hangup_disposition etc?
"Maybe you should beat around the bush less with your "requirements" for
your application you are expecting me to support for you."
I am just trying to gather statistics for my providers as I would with any
VoIP softswitch. (hangup causes per terminator per destination)
I don't think that this is a specific "application" rather than a general
necessity for VoIP carriers. It is also very useful for troubleshooting
purposes : when I look at my CDRs to find a call that I got a complain for,
I want to be able to tell if it was me or the provider who
hanged up and gave a specific hangup cause, so that I can troubleshoot the
issue better.
"Just be clear about what you want."
I want FS to reach that level of detailing and maturity in all aspects so
that it could be the softswitch of choice by any VoIP entrepreneur
(or hobbyist) and it is my strong belief that this can only be done by the
community giving feedback to the programmers about what
they find useful or not (i.e. experience from real-life situations). The
patches that you made the last few days were not intended for
me exclusively but for anyone that will face the same situations using FS.
If you want the community to stop sending feedback about
features/improvements you may as well close down this mailing list or just
use it as an announcement board.
I wish I was a c programmer and get involved with the project actively. But
I am not. And as far as I can tell most of the registered users
in this list aren't either. So they only way we can help is by testing and
suggesting.
Anthony Minessale wrote:
it is protocol neutral, that's why it starts with sip_
the variable can be any of:
send_bye
recv_bye
send_cancel
send_refuse
using that value you can determine the information you asked. I answered
your specific question which was:
determining "which side hanged up". Maybe you should beat around the bush
less with your "requirements" for your application you are expecting me to
support for you.
I already added 2 patches for you right. Just be clear about what you want.
On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs@kinetix.gr> (regs@kinetix.gr)
wrote:
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN:anthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
[url=sip:888@conference.freeswitch.org]sip:888@conference.freeswitch.org[/url]
iax:guest@conference.freeswitch.org/888 ([email]iax:guest@conference.freeswitch.org/888[/email])
googletalk:conf+888@conference.freeswitch.org ([email]googletalk:conf+888@conference.freeswitch.org[/email])
pstn:213-799-1400
________________________________
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr (regs@kinetix.gr)
-------------------------------------------
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
| --
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr (regs@kinetix.gr)
------------------------------------------- |
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sicfslist at gmail.com Guest
|
Posted: Tue Dec 09, 2008 11:16 am Post subject: [Freeswitch-users] Proto specific hangup cause issue |
|
|
Hello,
This is just my 2 cents ... but my experience has been that trying to catch all of the various variables (i.e. from XML_CDR) or otherwise can be a little trying (a row in your CDR database could be over 100 fields long!).
The best option here is to catch the UUID's for the 2 call legs, capture all SIP messaging, parse and dump the messaging, and then correlate the calls from the CDR from there.
Much easier than trying to do it from FS ... and most folks want to see SIP captures anyway (very broad set of tools to debug).
Measuring things like ASR, PDD, etc in my opinion is much easier from the raw messaging than trying to do something with FS CDR records.
On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos <regs@kinetix.gr (regs@kinetix.gr)> wrote:
Quote: |
We are currently in the migration process from our
current system to a FS based setup. We are in the process of
adapting our billing and routing to FS. All the CDRs (and variables)
related issues that we have been discussing on this mailing list
come from the need to extract the same level of information from FS as
we do with our current closed source proprietary system. So, we
chose FS because of the versatility it provides in every aspect (event
handling, config implementation etc.) and we strongly believe that all
these additions/fixes would be beneficial to many potential FS users.
We are at your disposal for more details in case you need
more information about what exactly we are trying to do. Basically,
our approach is from the VoIP carrier's point of view rather than the
PBX user's/implementor's. So, the details that we asked to be introduced
to FS come from real life issues that we have faced during the last few years
with various platforms and troubleshooting experiences with other VoIP carriers.
Michael Collins wrote: Quote: | Quote: | Thanks for your feedback. It definitely helps to know not only what
you need FS to do but why you need it to do so.
Do you have FS in production right now? Just curious.
Thanks,
MC
On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
<regs@kinetix.gr> (regs@kinetix.gr) wrote:
Quote: | "I already added 2 patches for you right. Just be clear about what you
want."
And I am grateful of that.
"it is protocol neutral, that's why it starts with sip_"
I didn't know that. I thought that the sip_ variables are protocol specific.
So one would expect there to be an iax_hangup_disposition,
woomera_hangup_disposition etc?
"Maybe you should beat around the bush less with your "requirements" for
your application you are expecting me to support for you."
I am just trying to gather statistics for my providers as I would with any
VoIP softswitch. (hangup causes per terminator per destination)
I don't think that this is a specific "application" rather than a general
necessity for VoIP carriers. It is also very useful for troubleshooting
purposes : when I look at my CDRs to find a call that I got a complain for,
I want to be able to tell if it was me or the provider who
hanged up and gave a specific hangup cause, so that I can troubleshoot the
issue better.
"Just be clear about what you want."
I want FS to reach that level of detailing and maturity in all aspects so
that it could be the softswitch of choice by any VoIP entrepreneur
(or hobbyist) and it is my strong belief that this can only be done by the
community giving feedback to the programmers about what
they find useful or not (i.e. experience from real-life situations). The
patches that you made the last few days were not intended for
me exclusively but for anyone that will face the same situations using FS.
If you want the community to stop sending feedback about
features/improvements you may as well close down this mailing list or just
use it as an announcement board.
I wish I was a c programmer and get involved with the project actively. But
I am not. And as far as I can tell most of the registered users
in this list aren't either. So they only way we can help is by testing and
suggesting.
Anthony Minessale wrote:
it is protocol neutral, that's why it starts with sip_
the variable can be any of:
send_bye
recv_bye
send_cancel
send_refuse
using that value you can determine the information you asked. I answered
your specific question which was:
determining "which side hanged up". Maybe you should beat around the bush
less with your "requirements" for your application you are expecting me to
support for you.
I already added 2 patches for you right. Just be clear about what you want.
On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs@kinetix.gr> (regs@kinetix.gr)
wrote:
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN:anthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888 ([email]iax:guest@conference.freeswitch.org/888[/email])
googletalk:conf+888@conference.freeswitch.org ([email]googletalk:conf+888@conference.freeswitch.org[/email])
pstn:213-799-1400
________________________________
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr (regs@kinetix.gr)
-------------------------------------------
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
| _______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
| |
Quote: | --
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr (regs@kinetix.gr)
------------------------------------------- |
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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regs at kinetix.gr Guest
|
Posted: Tue Dec 09, 2008 11:46 am Post subject: [Freeswitch-users] Proto specific hangup cause issue |
|
|
That approach introduces a third party application to
the setup (in order to capture and parse tha SIP messages)
that adds a lot in terms of complexity and reliability ( and cpu
usage). Also it could become a nightmare when you use a
mix of protocols (iax, sip, h323) and technologies (openzap etc).
In the case of a live debugging session, capturing is the most useful tool
but if you want to troubleshoot based on historical data (CDRs) then you
need some detailing. In addition you don't have to fill your databases
with all the fields that FS gives you in an XML cdr. You could
only pick those which are of interest in a particular application.
Shelby Ramsey wrote: Quote: | Hello,
This is just my 2 cents ... but my experience has been that trying to catch all of the various variables (i.e. from XML_CDR) or otherwise can be a little trying (a row in your CDR database could be over 100 fields long!).
The best option here is to catch the UUID's for the 2 call legs, capture all SIP messaging, parse and dump the messaging, and then correlate the calls from the CDR from there.
Much easier than trying to do it from FS ... and most folks want to see SIP captures anyway (very broad set of tools to debug).
Measuring things like ASR, PDD, etc in my opinion is much easier from the raw messaging than trying to do something with FS CDR records.
On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos <regs@kinetix.gr (regs@kinetix.gr)> wrote:
Quote: |
We are currently in the migration process from our
current system to a FS based setup. We are in the process of
adapting our billing and routing to FS. All the CDRs (and variables)
related issues that we have been discussing on this mailing list
come from the need to extract the same level of information from FS as
we do with our current closed source proprietary system. So, we
chose FS because of the versatility it provides in every aspect (event
handling, config implementation etc.) and we strongly believe that all
these additions/fixes would be beneficial to many potential FS users.
We are at your disposal for more details in case you need
more information about what exactly we are trying to do. Basically,
our approach is from the VoIP carrier's point of view rather than the
PBX user's/implementor's. So, the details that we asked to be introduced
to FS come from real life issues that we have faced during the last few years
with various platforms and troubleshooting experiences with other VoIP carriers.
Michael Collins wrote: Quote: | Quote: | Thanks for your feedback. It definitely helps to know not only what
you need FS to do but why you need it to do so.
Do you have FS in production right now? Just curious.
Thanks,
MC
On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
<regs@kinetix.gr> (regs@kinetix.gr) wrote:
Quote: | "I already added 2 patches for you right. Just be clear about what you
want."
And I am grateful of that.
"it is protocol neutral, that's why it starts with sip_"
I didn't know that. I thought that the sip_ variables are protocol specific.
So one would expect there to be an iax_hangup_disposition,
woomera_hangup_disposition etc?
"Maybe you should beat around the bush less with your "requirements" for
your application you are expecting me to support for you."
I am just trying to gather statistics for my providers as I would with any
VoIP softswitch. (hangup causes per terminator per destination)
I don't think that this is a specific "application" rather than a general
necessity for VoIP carriers. It is also very useful for troubleshooting
purposes : when I look at my CDRs to find a call that I got a complain for,
I want to be able to tell if it was me or the provider who
hanged up and gave a specific hangup cause, so that I can troubleshoot the
issue better.
"Just be clear about what you want."
I want FS to reach that level of detailing and maturity in all aspects so
that it could be the softswitch of choice by any VoIP entrepreneur
(or hobbyist) and it is my strong belief that this can only be done by the
community giving feedback to the programmers about what
they find useful or not (i.e. experience from real-life situations). The
patches that you made the last few days were not intended for
me exclusively but for anyone that will face the same situations using FS.
If you want the community to stop sending feedback about
features/improvements you may as well close down this mailing list or just
use it as an announcement board.
I wish I was a c programmer and get involved with the project actively. But
I am not. And as far as I can tell most of the registered users
in this list aren't either. So they only way we can help is by testing and
suggesting.
Anthony Minessale wrote:
it is protocol neutral, that's why it starts with sip_
the variable can be any of:
send_bye
recv_bye
send_cancel
send_refuse
using that value you can determine the information you asked. I answered
your specific question which was:
determining "which side hanged up". Maybe you should beat around the bush
less with your "requirements" for your application you are expecting me to
support for you.
I already added 2 patches for you right. Just be clear about what you want.
On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs@kinetix.gr>
wrote:
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org
pstn:213-799-1400
________________________________
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr
-------------------------------------------
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
| _______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
| |
Quote: | --
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr
------------------------------------------- |
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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regs at kinetix.gr Guest
|
Posted: Tue Dec 09, 2008 11:51 am Post subject: [Freeswitch-users] Proto specific hangup cause issue |
|
|
That approach introduces a third party application to
the setup (in order to capture and parse tha SIP messages)
that adds a lot in terms of complexity and reliability ( and cpu
usage). Also it could become a nightmare when you use a
mix of protocols (iax, sip, h323) and technologies (openzap etc).
In the case of a live debugging session, capturing is the most useful tool
but if you want to troubleshoot based on historical data (CDRs) then you
need some detailing. In addition you don't have to fill your databases
with all the fields that FS gives you in an XML cdr. You could
only pick those which are of interest in a particular application.
Shelby Ramsey wrote: Quote: | Hello,
This is just my 2 cents ... but my experience has been that trying to catch all of the various variables (i.e. from XML_CDR) or otherwise can be a little trying (a row in your CDR database could be over 100 fields long!).
The best option here is to catch the UUID's for the 2 call legs, capture all SIP messaging, parse and dump the messaging, and then correlate the calls from the CDR from there.
Much easier than trying to do it from FS ... and most folks want to see SIP captures anyway (very broad set of tools to debug).
Measuring things like ASR, PDD, etc in my opinion is much easier from the raw messaging than trying to do something with FS CDR records.
On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos <regs@kinetix.gr (regs@kinetix.gr)> wrote:
Quote: |
We are currently in the migration process from our
current system to a FS based setup. We are in the process of
adapting our billing and routing to FS. All the CDRs (and variables)
related issues that we have been discussing on this mailing list
come from the need to extract the same level of information from FS as
we do with our current closed source proprietary system. So, we
chose FS because of the versatility it provides in every aspect (event
handling, config implementation etc.) and we strongly believe that all
these additions/fixes would be beneficial to many potential FS users.
We are at your disposal for more details in case you need
more information about what exactly we are trying to do. Basically,
our approach is from the VoIP carrier's point of view rather than the
PBX user's/implementor's. So, the details that we asked to be introduced
to FS come from real life issues that we have faced during the last few years
with various platforms and troubleshooting experiences with other VoIP carriers.
Michael Collins wrote: Quote: | Quote: | Thanks for your feedback. It definitely helps to know not only what
you need FS to do but why you need it to do so.
Do you have FS in production right now? Just curious.
Thanks,
MC
On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
<regs@kinetix.gr> (regs@kinetix.gr) wrote:
Quote: | "I already added 2 patches for you right. Just be clear about what you
want."
And I am grateful of that.
"it is protocol neutral, that's why it starts with sip_"
I didn't know that. I thought that the sip_ variables are protocol specific.
So one would expect there to be an iax_hangup_disposition,
woomera_hangup_disposition etc?
"Maybe you should beat around the bush less with your "requirements" for
your application you are expecting me to support for you."
I am just trying to gather statistics for my providers as I would with any
VoIP softswitch. (hangup causes per terminator per destination)
I don't think that this is a specific "application" rather than a general
necessity for VoIP carriers. It is also very useful for troubleshooting
purposes : when I look at my CDRs to find a call that I got a complain for,
I want to be able to tell if it was me or the provider who
hanged up and gave a specific hangup cause, so that I can troubleshoot the
issue better.
"Just be clear about what you want."
I want FS to reach that level of detailing and maturity in all aspects so
that it could be the softswitch of choice by any VoIP entrepreneur
(or hobbyist) and it is my strong belief that this can only be done by the
community giving feedback to the programmers about what
they find useful or not (i.e. experience from real-life situations). The
patches that you made the last few days were not intended for
me exclusively but for anyone that will face the same situations using FS.
If you want the community to stop sending feedback about
features/improvements you may as well close down this mailing list or just
use it as an announcement board.
I wish I was a c programmer and get involved with the project actively. But
I am not. And as far as I can tell most of the registered users
in this list aren't either. So they only way we can help is by testing and
suggesting.
Anthony Minessale wrote:
it is protocol neutral, that's why it starts with sip_
the variable can be any of:
send_bye
recv_bye
send_cancel
send_refuse
using that value you can determine the information you asked. I answered
your specific question which was:
determining "which side hanged up". Maybe you should beat around the bush
less with your "requirements" for your application you are expecting me to
support for you.
I already added 2 patches for you right. Just be clear about what you want.
On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs@kinetix.gr>
wrote:
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org
pstn:213-799-1400
________________________________
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr
-------------------------------------------
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
| _______________________________________________
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
| |
Quote: | --
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr
------------------------------------------- |
_______________________________________________
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| |
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Back to top |
|
|
vhatz at kinetix.gr Guest
|
Posted: Tue Dec 09, 2008 11:52 am Post subject: [Freeswitch-users] Proto specific hangup cause issue |
|
|
Shelby Ramsey wrote:
Quote: | Hello,
This is just my 2 cents ... but my experience has been that trying to
catch all of the various variables (i.e. from XML_CDR) or otherwise can
be a little trying (a row in your CDR database could be over 100 fields
long!).
The best option here is to catch the UUID's for the 2 call legs, capture
all SIP messaging, parse and dump the messaging, and then correlate the
calls from the CDR from there.
Much easier than trying to do it from FS ... and most folks want to see
SIP captures anyway (very broad set of tools to debug).
Measuring things like ASR, PDD, etc in my opinion is much easier from
the raw messaging than trying to do something with FS CDR records.
|
That can certainly be an option, especially for debugging purposes.
However, under heavy load (imagine a few thousands of calls per hour, a
few millions per day) logging and parsing all the SIP messages on file
will be a problem.
Also, logging SIP messages is oriented to SIP only, when a more protocol
agnostic approach could be followed. Plus, we would still need to parse
a lot of text to extract the information that we need, while in a CDR
(even a long one with many fields) we have a lot of information with a
minimum hassle.
We've seen in production environments that excessive logging wastes I/O
power and disk space, this is why (we at least) turn it on in our
various systems only when we need it for troubleshooting, and
immediately turn it off afterwards.
Additionally, a very long CDR is a lot less text to write on disk once,
after the call is over, rather than writing many, whole packets during
the duration of a call.
A 100-field-CDR on file could not be much of a problem, because usually
these the raw CDR fields are rarely imported in a database in their
entirety for billing or QoS analysis. A lot of the information which is
not used directly & immediately for billing or QoS analysis remains on
file in case needs to do basic troubleshooting in arrears. Granted, we
would not have the same amount of information as with the written SIP
messages, but it is useful nonetheless.
Of course, I need to stress that I write all this coming from the
background of VoIP carriers. The above could apply well for typical &
simple scenarios, where a call leg comes into FS and another calls leg
comes out of it, which is what most carriers do.
If we need billing and QoS analysis for IVR's, queues, call transfers,
etc, then yes, one-line CDRs would not do. In this case, logging whole
packets could be a solution, although an event-based approach could be
much better to cover all protocols/technologies (IAX, TDM cards, etc), IMHO.
Best regards,
Vlasis Hatzistavrou
Kinetix Tele.com Hellas Ltd.
Monastiriou 9 & Enotikon
54627
Thessaloniki
Greece
Tel.: +302310556134
Fax: +302310556134 (ext. 0)
GSM: +306977835653
e-mail: vhatz@kinetix.gr
http://www.kinetix.gr
Quote: |
On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos <regs@kinetix.gr
<mailto:regs@kinetix.gr>> wrote:
We are currently in the migration process from our
current system to a FS based setup. We are in the process of
adapting our billing and routing to FS. All the CDRs (and variables)
related issues that we have been discussing on this mailing list
come from the need to extract the same level of information from FS as
we do with our current closed source proprietary system. So, we
chose FS because of the versatility it provides in every aspect (event
handling, config implementation etc.) and we strongly believe that all
these additions/fixes would be beneficial to many potential FS users.
We are at your disposal for more details in case you need
more information about what exactly we are trying to do. Basically,
our approach is from the VoIP carrier's point of view rather than the
PBX user's/implementor's. So, the details that we asked to be introduced
to FS come from real life issues that we have faced during the last
few years
with various platforms and troubleshooting experiences with other
VoIP carriers.
Michael Collins wrote:
Quote: | Thanks for your feedback. It definitely helps to know not only what
you need FS to do but why you need it to do so.
Do you have FS in production right now? Just curious.
Thanks,
MC
On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
<regs@kinetix.gr> <mailto:regs@kinetix.gr> wrote:
Quote: | "I already added 2 patches for you right. Just be clear about what you
want."
And I am grateful of that.
"it is protocol neutral, that's why it starts with sip_"
I didn't know that. I thought that the sip_ variables are protocol specific.
So one would expect there to be an iax_hangup_disposition,
woomera_hangup_disposition etc?
"Maybe you should beat around the bush less with your "requirements" for
your application you are expecting me to support for you."
I am just trying to gather statistics for my providers as I would with any
VoIP softswitch. (hangup causes per terminator per destination)
I don't think that this is a specific "application" rather than a general
necessity for VoIP carriers. It is also very useful for troubleshooting
purposes : when I look at my CDRs to find a call that I got a complain for,
I want to be able to tell if it was me or the provider who
hanged up and gave a specific hangup cause, so that I can troubleshoot the
issue better.
"Just be clear about what you want."
I want FS to reach that level of detailing and maturity in all aspects so
that it could be the softswitch of choice by any VoIP entrepreneur
(or hobbyist) and it is my strong belief that this can only be done by the
community giving feedback to the programmers about what
they find useful or not (i.e. experience from real-life situations). The
patches that you made the last few days were not intended for
me exclusively but for anyone that will face the same situations using FS.
If you want the community to stop sending feedback about
features/improvements you may as well close down this mailing list or just
use it as an announcement board.
I wish I was a c programmer and get involved with the project actively. But
I am not. And as far as I can tell most of the registered users
in this list aren't either. So they only way we can help is by testing and
suggesting.
Anthony Minessale wrote:
it is protocol neutral, that's why it starts with sip_
the variable can be any of:
send_bye
recv_bye
send_cancel
send_refuse
using that value you can determine the information you asked. I answered
your specific question which was:
determining "which side hanged up". Maybe you should beat around the bush
less with your "requirements" for your application you are expecting me to
support for you.
I already added 2 patches for you right. Just be clear about what you want.
On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs@kinetix.gr> <mailto:regs@kinetix.gr>
wrote:
Quote: | Not necessarily. For instance I got a "send_cancel" when the
calling party hanged up before the other party could pick up.
Also, shouldn't something like that be protocol/technology
neutral?
Anthony Minessale wrote:
sip_hangup_disposition will be set to recv_bye on the side that was
hungup.
On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <regs@kinetix.gr> <mailto:regs@kinetix.gr>
wrote:
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com <mailto:MSN:anthony_minessale@hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com>
IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888 <mailto:iax:guest@conference.freeswitch.org/888>
googletalk:conf+888@conference.freeswitch.org <mailto:googletalk:conf+888@conference.freeswitch.org>
pstn:213-799-1400
________________________________
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr <mailto:regs@kinetix.gr>
-------------------------------------------
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
| --
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com <mailto:MSN:anthony_minessale@hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com>
IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888 <mailto:iax:guest@conference.freeswitch.org/888>
googletalk:conf+888@conference.freeswitch.org <mailto:googletalk:conf+888@conference.freeswitch.org>
pstn:213-799-1400
________________________________
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr <mailto:regs@kinetix.gr>
-------------------------------------------
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
| _______________________________________________
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr <mailto:regs@kinetix.gr>
-------------------------------------------
_______________________________________________
Freeswitch-users mailing list
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|
Back to top |
|
|
regs at kinetix.gr Guest
|
Posted: Tue Dec 09, 2008 11:55 am Post subject: [Freeswitch-users] Proto specific hangup cause issue |
|
|
That approach introduces a third party application to
the setup (in order to capture and parse tha SIP messages)
that adds a lot in terms of complexity and reliability ( and cpu
usage). Also it could become a nightmare when you use a
mix of protocols (iax, sip, h323) and technologies (openzap etc).
In the case of a live debugging session, capturing is the most useful tool
but if you want to troubleshoot based on historical data (CDRs) then you
need some detailing. In addition you don't have to fill your databases
with all the fields that FS gives you in an XML cdr. You could
only pick those which are of interest in a particular application.
Shelby Ramsey wrote: Quote: | Hello,
This is just my 2 cents ... but my experience has been that trying to catch all of the various variables (i.e. from XML_CDR) or otherwise can be a little trying (a row in your CDR database could be over 100 fields long!).
The best option here is to catch the UUID's for the 2 call legs, capture all SIP messaging, parse and dump the messaging, and then correlate the calls from the CDR from there.
Much easier than trying to do it from FS ... and most folks want to see SIP captures anyway (very broad set of tools to debug).
Measuring things like ASR, PDD, etc in my opinion is much easier from the raw messaging than trying to do something with FS CDR records.
On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos <regs@kinetix.gr (regs@kinetix.gr)> wrote:
Quote: |
We are currently in the migration process from our
current system to a FS based setup. We are in the process of
adapting our billing and routing to FS. All the CDRs (and variables)
related issues that we have been discussing on this mailing list
come from the need to extract the same level of information from FS as
we do with our current closed source proprietary system. So, we
chose FS because of the versatility it provides in every aspect (event
handling, config implementation etc.) and we strongly believe that all
these additions/fixes would be beneficial to many potential FS users.
We are at your disposal for more details in case you need
more information about what exactly we are trying to do. Basically,
our approach is from the VoIP carrier's point of view rather than the
PBX user's/implementor's. So, the details that we asked to be introduced
to FS come from real life issues that we have faced during the last few years
with various platforms and troubleshooting experiences with other VoIP carriers.
Michael Collins wrote: Quote: | Quote: | Thanks for your feedback. It definitely helps to know not only what
you need FS to do but why you need it to do so.
Do you have FS in production right now? Just curious.
Thanks,
MC
On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
<regs@kinetix.gr> (regs@kinetix.gr) wrote:
Quote: | "I already added 2 patches for you right. Just be clear about what you
want."
And I am grateful of that.
"it is protocol neutral, that's why it starts with sip_"
I didn't know that. I thought that the sip_ variables are protocol specific.
So one would expect there to be an iax_hangup_disposition,
woomera_hangup_disposition etc?
"Maybe you should beat around the bush less with your "requirements" for
your application you are expecting me to support for you."
I am just trying to gather statistics for my providers as I would with any
VoIP softswitch. (hangup causes per terminator per destination)
I don't think that this is a specific "application" rather than a general
necessity for VoIP carriers. It is also very useful for troubleshooting
purposes : when I look at my CDRs to find a call that I got a complain for,
I want to be able to tell if it was me or the provider who
hanged up and gave a specific hangup cause, so that I can troubleshoot the
issue better.
"Just be clear about what you want."
I want FS to reach that level of detailing and maturity in all aspects so
that it could be the softswitch of choice by any VoIP entrepreneur
(or hobbyist) and it is my strong belief that this can only be done by the
community giving feedback to the programmers about what
they find useful or not (i.e. experience from real-life situations). The
patches that you made the last few days were not intended for
me exclusively but for anyone that will face the same situations using FS.
If you want the community to stop sending feedback about
features/improvements you may as well close down this mailing list or just
use it as an announcement board.
I wish I was a c programmer and get involved with the project actively. But
I am not. And as far as I can tell most of the registered users
in this list aren't either. So they only way we can help is by testing and
suggesting.
Anthony Minessale wrote:
it is protocol neutral, that's why it starts with sip_
the variable can be any of:
send_bye
recv_bye
send_cancel
send_refuse
using that value you can determine the information you asked. I answered
your specific question which was:
determining "which side hanged up". Maybe you should beat around the bush
less with your "requirements" for your application you are expecting me to
support for you.
I already added 2 patches for you right. Just be clear about what you want.
On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs@kinetix.gr>
wrote:
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org
pstn:213-799-1400
________________________________
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr
-------------------------------------------
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
| _______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
| |
Quote: | --
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr
------------------------------------------- |
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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| |
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|
Back to top |
|
|
anthony.minessale at g... Guest
|
Posted: Tue Dec 09, 2008 12:21 pm Post subject: [Freeswitch-users] Proto specific hangup cause issue |
|
|
see this is better.
That's why I asked you to be more specific about what you want because the tiny back and
forth questions were not exposing your intent or needs at all. I answer every email I can and when threads start to grow without getting to the point i start to get frustrated.
Now that you have opened up the discussion you have more people chiming in on the topic.
Yes the sip_* variables are unique to SIP and the one that says proto_specific are all
done per implementation.
If you would like to suggest a list of standard variables you think apply to all calls
or something you feel is missing, we can consider injecting them into the code.
On Tue, Dec 9, 2008 at 10:42 AM, Vlasis Hatzistavrou (KTI) <vhatz@kinetix.gr (vhatz@kinetix.gr)> wrote:
Quote: | Shelby Ramsey wrote:
Quote: | Hello,
This is just my 2 cents ... but my experience has been that trying to
catch all of the various variables (i.e. from XML_CDR) or otherwise can
be a little trying (a row in your CDR database could be over 100 fields
long!).
The best option here is to catch the UUID's for the 2 call legs, capture
all SIP messaging, parse and dump the messaging, and then correlate the
calls from the CDR from there.
Much easier than trying to do it from FS ... and most folks want to see
SIP captures anyway (very broad set of tools to debug).
Measuring things like ASR, PDD, etc in my opinion is much easier from
the raw messaging than trying to do something with FS CDR records.
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That can certainly be an option, especially for debugging purposes.
However, under heavy load (imagine a few thousands of calls per hour, a
few millions per day) logging and parsing all the SIP messages on file
will be a problem.
Also, logging SIP messages is oriented to SIP only, when a more protocol
agnostic approach could be followed. Plus, we would still need to parse
a lot of text to extract the information that we need, while in a CDR
(even a long one with many fields) we have a lot of information with a
minimum hassle.
We've seen in production environments that excessive logging wastes I/O
power and disk space, this is why (we at least) turn it on in our
various systems only when we need it for troubleshooting, and
immediately turn it off afterwards.
Additionally, a very long CDR is a lot less text to write on disk once,
after the call is over, rather than writing many, whole packets during
the duration of a call.
A 100-field-CDR on file could not be much of a problem, because usually
these the raw CDR fields are rarely imported in a database in their
entirety for billing or QoS analysis. A lot of the information which is
not used directly & immediately for billing or QoS analysis remains on
file in case needs to do basic troubleshooting in arrears. Granted, we
would not have the same amount of information as with the written SIP
messages, but it is useful nonetheless.
Of course, I need to stress that I write all this coming from the
background of VoIP carriers. The above could apply well for typical &
simple scenarios, where a call leg comes into FS and another calls leg
comes out of it, which is what most carriers do.
If we need billing and QoS analysis for IVR's, queues, call transfers,
etc, then yes, one-line CDRs would not do. In this case, logging whole
packets could be a solution, although an event-based approach could be
much better to cover all protocols/technologies (IAX, TDM cards, etc), IMHO.
Best regards,
Vlasis Hatzistavrou
Kinetix Tele.com Hellas Ltd.
Monastiriou 9 & Enotikon
54627
Thessaloniki
Greece
Tel.: +302310556134
Fax: +302310556134 (ext. 0)
GSM: +306977835653
e-mail: vhatz@kinetix.gr (vhatz@kinetix.gr)
http://www.kinetix.gr
Quote: |
On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos <regs@kinetix.gr (regs@kinetix.gr)
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Quote: | <mailto:regs@kinetix.gr (regs@kinetix.gr)>> wrote:
We are currently in the migration process from our
current system to a FS based setup. We are in the process of
adapting our billing and routing to FS. All the CDRs (and variables)
related issues that we have been discussing on this mailing list
come from the need to extract the same level of information from FS as
we do with our current closed source proprietary system. So, we
chose FS because of the versatility it provides in every aspect (event
handling, config implementation etc.) and we strongly believe that all
these additions/fixes would be beneficial to many potential FS users.
We are at your disposal for more details in case you need
more information about what exactly we are trying to do. Basically,
our approach is from the VoIP carrier's point of view rather than the
PBX user's/implementor's. So, the details that we asked to be introduced
to FS come from real life issues that we have faced during the last
few years
with various platforms and troubleshooting experiences with other
VoIP carriers.
Michael Collins wrote:
Quote: | Thanks for your feedback. It definitely helps to know not only what
you need FS to do but why you need it to do so.
Do you have FS in production right now? Just curious.
Thanks,
MC
On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
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Quote: | Quote: | <regs@kinetix.gr (regs@kinetix.gr)> <mailto:regs@kinetix.gr (regs@kinetix.gr)> wrote:
Quote: | "I already added 2 patches for you right. Just be clear about what you
want."
And I am grateful of that.
"it is protocol neutral, that's why it starts with sip_"
I didn't know that. I thought that the sip_ variables are protocol specific.
So one would expect there to be an iax_hangup_disposition,
woomera_hangup_disposition etc?
"Maybe you should beat around the bush less with your "requirements" for
your application you are expecting me to support for you."
I am just trying to gather statistics for my providers as I would with any
VoIP softswitch. (hangup causes per terminator per destination)
I don't think that this is a specific "application" rather than a general
necessity for VoIP carriers. It is also very useful for troubleshooting
purposes : when I look at my CDRs to find a call that I got a complain for,
I want to be able to tell if it was me or the provider who
hanged up and gave a specific hangup cause, so that I can troubleshoot the
issue better.
"Just be clear about what you want."
I want FS to reach that level of detailing and maturity in all aspects so
that it could be the softswitch of choice by any VoIP entrepreneur
(or hobbyist) and it is my strong belief that this can only be done by the
community giving feedback to the programmers about what
they find useful or not (i.e. experience from real-life situations). The
patches that you made the last few days were not intended for
me exclusively but for anyone that will face the same situations using FS.
If you want the community to stop sending feedback about
features/improvements you may as well close down this mailing list or just
use it as an announcement board.
I wish I was a c programmer and get involved with the project actively. But
I am not. And as far as I can tell most of the registered users
in this list aren't either. So they only way we can help is by testing and
suggesting.
Anthony Minessale wrote:
it is protocol neutral, that's why it starts with sip_
the variable can be any of:
send_bye
recv_bye
send_cancel
send_refuse
using that value you can determine the information you asked. I answered
your specific question which was:
determining "which side hanged up". Maybe you should beat around the bush
less with your "requirements" for your application you are expecting me to
support for you.
I already added 2 patches for you right. Just be clear about what you want.
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Quote: | Quote: | Quote: | On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs@kinetix.gr (regs@kinetix.gr)> <mailto:regs@kinetix.gr (regs@kinetix.gr)>
wrote:
Quote: | Not necessarily. For instance I got a "send_cancel" when the
calling party hanged up before the other party could pick up.
Also, shouldn't something like that be protocol/technology
neutral?
Anthony Minessale wrote:
sip_hangup_disposition will be set to recv_bye on the side that was
hungup.
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Quote: | Quote: | Quote: | Quote: | On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <regs@kinetix.gr (regs@kinetix.gr)> <mailto:regs@kinetix.gr (regs@kinetix.gr)>
wrote:
Quote: | Any updates about the "which side hanged up" potential variable?
Michael S Collins wrote:
Makes sense. I will look into this.
-MC
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Quote: | Quote: | Quote: | Quote: | Quote: | On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <regs@kinetix.gr (regs@kinetix.gr)> <mailto:regs@kinetix.gr (regs@kinetix.gr)>
wrote:
I am sending this second email to request/suggest/enquire about something
relevant :
Wouldn't it be useful to know which end of a specific call leg send the
protocol
specific hangup cause? Otherwise it would be difficult to understand what
really happened.
Michael S Collins wrote:
I will do some research on this and let you know what I find out.
Question: are these internal calls or pstn or ?? Just curious about
your environment.
Thanks,
MC
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Quote: | Quote: | Quote: | Quote: | Quote: | On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs@kinetix.gr (regs@kinetix.gr)> <mailto:regs@kinetix.gr (regs@kinetix.gr)>
wrote:
The proto_specific_hangup_cause is missing on one of the two
call legs. When the caller hangs up it is missing from the a-leg CDR.
When the callee hangs up it is missing from the b-leg CDR. Is this
nornal?
And another question : what piece of info could inform me about who
hanged up?
--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
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Quote: | Quote: | Quote: | Quote: | MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email]) <mailto:MSN ([email]MSN[/email]):anthony_minessale@hotmail.com (anthony_minessale@hotmail.com)>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email]) <mailto:GTALK ([email]GTALK[/email])/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])>
IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
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Quote: | Quote: | Quote: | Quote: | iax ([email]iax[/email]):guest@conference.freeswitch.org/888>
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email]) <mailto:googletalk ([email]googletalk[/email]):conf+888@conference.freeswitch.org ([email]conf%2B888@conference.freeswitch.org[/email])>
pstn:213-799-1400
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Freeswitch-users mailing list
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Quote: | Quote: | Quote: | MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email]) <mailto:MSN ([email]MSN[/email]):anthony_minessale@hotmail.com (anthony_minessale@hotmail.com)>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email]) <mailto:GTALK ([email]GTALK[/email])/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])>
IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
|
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Quote: | Quote: | Quote: | iax ([email]iax[/email]):guest@conference.freeswitch.org/888>
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email]) <mailto:googletalk ([email]googletalk[/email]):conf+888@conference.freeswitch.org ([email]conf%2B888@conference.freeswitch.org[/email])>
pstn:213-799-1400
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Freeswitch-users mailing list
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Quote: | Quote: | _______________________________________________
Freeswitch-users mailing list
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Quote: |
--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
|
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400 |
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